What do you see when you run asterisk –r and dial 210 or 211 from one of the phones
James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by "reply to sender only" message and destroy all electronic and hard copies of the communication, including attachments. "Common sense is the collection of prejudices acquired by age eighteen." -- Albert Einstein "Once you can accept the universe as matter expanding into nothing that is something,wearing stripes with plaid comes easy." -- Albert Einstein "I know a little of everything, but a lot of nothing" From: [email protected] [mailto:[email protected]] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 11:19 AM To: [email protected] Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [r...@asterisk asterisk]# cat sip.conf [general] bindport=5060 bindaddr = 0.0.0.0 [BT201] type=friend context=intern host=192.168.4.210 secret=testpaswoord [GXP1200] type=friend context=intern host=192.168.4.211 secret=testpaswoord extensions.conf : [r...@asterisk asterisk]# cat extensions.conf [intern] exten => 210,1,Dial(SIP/BT201) exten => 211,1,Dial(SIP/GXP1200) Asterisk CLI shows me : asterisk*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status GXP1200 192.168.4.211 5060 Unmonitored BT201 192.168.4.210 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] asterisk*CLI> dialplan show intern [ Context 'intern' created by 'pbx_config' ] '210' => 1. Dial(SIP/BT201) [pbx_config] '211' => 1. Dial(SIP/GXP1200) [pbx_config] I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas.
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